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32 #include "config_components.h"
71 uint8_t audio_buf[4][8192];
86 if (shift < 0x2 || shift > 0xd) {
88 }
else if (
shift < 0x8) {
99 static const uint8_t *dv_extract_pack(
const uint8_t *
frame,
enum DVPackType t)
104 for (
c = 0;
c < 10;
c++) {
107 if (
c&1) offs = (80 * 6 + 80 * 16 * 0 + 3 +
c*12000);
108 else offs = (80 * 6 + 80 * 16 * 3 + 3 +
c*12000);
111 if (
c&1) offs = (80 * 6 + 80 * 16 * 1 + 3 +
c*12000);
112 else offs = (80 * 6 + 80 * 16 * 4 + 3 +
c*12000);
115 if (
c&1) offs = (80 * 3 + 8 +
c*12000);
116 else offs = (80 * 5 + 48 + 5 +
c*12000);
119 offs = (80*1 + 3 + 3);
124 if (
frame[offs] == t)
131 static const int dv_audio_frequency[3] = {
143 static int dv_extract_audio(
const uint8_t *
frame, uint8_t **ppcm,
146 int size, chan,
i, j, d, of, smpls, freq,
quant, half_ch;
148 const uint8_t *as_pack;
155 smpls = as_pack[1] & 0x3f;
156 freq = as_pack[4] >> 3 & 0x07;
157 quant = as_pack[4] & 0x07;
170 ipcm = (sys->
height == 720 && !(
frame[1] & 0x0C)) ? 2 : 0;
178 for (chan = 0; chan < sys->
n_difchan; chan++) {
187 if (
quant == 1 &&
i == half_ch) {
196 for (j = 0; j < 9; j++) {
197 for (d = 8; d < 80; d += 2) {
206 pcm[of * 2] =
frame[d + 1];
207 pcm[of * 2 + 1] =
frame[d];
209 if (pcm[of * 2 + 1] == 0x80 && pcm[of * 2] == 0x00)
212 lc = ((uint16_t)
frame[d] << 4) |
213 ((uint16_t)
frame[d + 2] >> 4);
214 rc = ((uint16_t)
frame[d + 1] << 4) |
215 ((uint16_t)
frame[d + 2] & 0x0f);
226 pcm[of * 2] = lc & 0xff;
227 pcm[of * 2 + 1] = lc >> 8;
232 pcm[of * 2] = rc & 0xff;
233 pcm[of * 2 + 1] = rc >> 8;
248 const uint8_t *as_pack;
249 int freq, stype, smpls,
quant,
i, ach, sr;
252 if (!as_pack || !
c->sys) {
257 smpls = as_pack[1] & 0x3f;
258 freq = as_pack[4] >> 3 & 0x07;
259 stype = as_pack[3] & 0x1f;
260 quant = as_pack[4] & 0x07;
264 "Unrecognized audio sample rate index (%d)\n", freq);
267 sr = dv_audio_frequency[freq];
276 ach = ((
int[4]) { 1, 0, 2, 4 })[stype];
277 if (ach == 1 &&
quant && freq == 2)
282 for (
i = 0;
i < ach;
i++) {
292 c->ast[
i]->start_time = 0;
293 c->ast[
i]->codecpar->bit_rate = 2 * sr * 16;
295 c->ast[
i]->codecpar->sample_rate = sr;
296 c->audio_pkt[
i].last_sample_rate = sr;
298 c->audio_pkt[
i].size = 0;
299 c->audio_pkt[
i].data =
c->audio_buf[
i];
300 c->audio_pkt[
i].stream_index =
c->ast[
i]->index;
303 c->audio_pkt[
i].duration = 0;
304 c->audio_pkt[
i].pos = -1;
307 c->audio_pkt[
i].sample_rate = sr;
311 return (
c->sys->audio_min_samples[freq] + smpls) * 4;
316 const uint8_t *vsc_pack;
320 par =
c->vst->codecpar;
322 c->vst->avg_frame_rate =
av_inv_q(
c->vst->time_base);
326 apt =
frame[4] & 0x07;
327 is16_9 = (vsc_pack && ((vsc_pack[2] & 0x07) == 0x02 ||
328 (!apt && (vsc_pack[2] & 0x07) == 0x07)));
329 c->vst->sample_aspect_ratio =
c->sys->sar[is16_9];
333 return c->sys->frame_size;
338 const uint8_t *tc_pack;
343 int prevent_df =
c->sys->ltc_divisor == 25 ||
c->sys->ltc_divisor == 50;
363 c->vst->start_time = 0;
381 if (dv_init_demux(
s,
c)) {
394 for (
i = 0;
i <
c->ach;
i++) {
395 if (
c->ast[
i] &&
c->audio_pkt[
i].size) {
409 if (dpkt->last_sample_rate != dpkt->sample_rate) {
413 dpkt->last_sample_rate = dpkt->sample_rate;
428 uint8_t *ppcm[5] = { 0 };
438 size = dv_extract_audio_info(
c,
buf);
444 c->ast[0]->time_base);
448 c->ast[0]->time_base);
454 next_pts_video :
c->next_pts_audio;
459 for (
i = 0;
i <
c->ach;
i++) {
467 ppcm[
i] =
c->audio_buf[
i];
470 dv_extract_audio(
buf, ppcm,
c->sys);
474 if (
c->sys->height == 720) {
476 c->audio_pkt[2].size =
c->audio_pkt[3].size = 0;
478 c->audio_pkt[0].size =
c->audio_pkt[1].size = 0;
484 size = dv_extract_video_info(
c,
buf);
527 c->frames = !
c->sys ? 0 :
529 c->next_pts_video = ts;
533 c->audio_pkt[0].size =
c->audio_pkt[1].size = 0;
534 c->audio_pkt[2].size =
c->audio_pkt[3].size = 0;
541 typedef struct RawDVContext {
552 #define PARTIAL_FRAME_SIZE (3 * 80)
553 uint8_t partial_frame[PARTIAL_FRAME_SIZE];
554 RawDVContext *
c =
s->priv_data;
560 if (
ret < PARTIAL_FRAME_SIZE) {
565 ret = dv_extract_timecode(&
c->dv_demux, partial_frame, timecode);
578 unsigned state, marker_pos = 0;
579 RawDVContext *
c =
s->priv_data;
583 if ((
ret = dv_init_demux(
s, &
c->dv_demux)) < 0)
587 while ((
state & 0xffffff7f) != 0x1f07003f) {
592 if (
state == 0x003f0700 ||
state == 0xff3f0700)
611 if (!
c->dv_demux.sys) {
613 "Can't determine profile of DV input stream.\n");
619 c->dv_demux.sys->time_base);
630 RawDVContext *
c =
s->priv_data;
637 if (!
c->dv_demux.sys)
639 size =
c->dv_demux.sys->frame_size;
643 }
else if (
ret == 0) {
656 RawDVContext *
r =
s->priv_data;
661 if (stream_index !=
c->vst->index)
662 timestamp =
av_rescale_q(timestamp,
s->streams[stream_index]->time_base,
665 offset = dv_frame_offset(
s,
c, ×tamp);
676 unsigned marker_pos = 0;
680 int secondary_matches = 0;
685 for (
i = 0;
i <
p->buf_size-4;
i++) {
687 if ((
state & 0x0007f840) == 0x00070000) {
690 if ((
state & 0xff07ff7f) == 0x1f07003f) {
692 if ((
state & 0xffffff7f) == 0x1f07003f) {
698 if (
state == 0x003f0700 ||
state == 0xff3f0700)
700 if (
state == 0xff3f0701 &&
i - marker_pos == 80)
705 if (matches &&
p->buf_size / matches < 1024 * 1024) {
706 if (matches > 4 || firstmatch ||
707 (secondary_matches >= 10 &&
708 p->buf_size / secondary_matches < 24000))
719 .p.extensions =
"dv,dif",
720 .priv_data_size =
sizeof(RawDVContext),
727 #else // CONFIG_DV_DEMUXER
743 #endif // CONFIG_DV_DEMUXER
#define AV_TIMECODE_STR_SIZE
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later. That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another. Frame references ownership and permissions
#define AV_CHANNEL_LAYOUT_STEREO
This struct describes the properties of an encoded stream.
AVStream * avformat_new_stream(AVFormatContext *s, const struct AVCodec *c)
Add a new stream to a media file.
For DV, one packet corresponds exactly to one frame.
int64_t duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
int64_t avio_size(AVIOContext *s)
Get the filesize.
#define AV_PKT_FLAG_KEY
The packet contains a keyframe.
static int read_seek(AVFormatContext *ctx, int stream_index, int64_t timestamp, int flags)
if it could not because there are no more frames
static av_always_inline int64_t avio_tell(AVIOContext *s)
ftell() equivalent for AVIOContext.
#define DV_TIMESCALE_AUDIO
static const uint8_t quant[64]
unsigned int avio_rb32(AVIOContext *s)
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
#define FF_ARRAY_ELEMS(a)
static int read_packet(void *opaque, uint8_t *buf, int buf_size)
#define av_assert0(cond)
assert() equivalent, that is always enabled.
int64_t data_offset
offset of the first packet
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
#define DV_TIMESCALE_VIDEO
and forward the result(frame or status change) to the corresponding input. If nothing is possible
Rational number (pair of numerator and denominator).
This structure contains the data a format has to probe a file.
Undefined Behavior In the C some operations are like signed integer dereferencing freed accessing outside allocated Undefined Behavior must not occur in a C it is not safe even if the output of undefined operations is unused The unsafety may seem nit picking but Optimizing compilers have in fact optimized code on the assumption that no undefined Behavior occurs Optimizing code based on wrong assumptions can and has in some cases lead to effects beyond the output of computations The signed integer overflow problem in speed critical code Code which is highly optimized and works with signed integers sometimes has the problem that often the output of the computation does not c
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
An AVChannelLayout holds information about the channel layout of audio data.
static int shift(int a, int b)
const AVDVProfile * av_dv_frame_profile(const AVDVProfile *sys, const uint8_t *frame, unsigned buf_size)
Get a DV profile for the provided compressed frame.
int ff_add_param_change(AVPacket *pkt, int32_t channels, uint64_t channel_layout, int32_t sample_rate, int32_t width, int32_t height)
Add side data to a packet for changing parameters to the given values.
static uint16_t dv_audio_12to16(uint16_t sample)
#define AV_NOPTS_VALUE
Undefined timestamp value.
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_WL16 uint64_t_TMPL AV_WB64 unsigned int_TMPL AV_RB32
#define DV_MAX_FRAME_SIZE
largest possible DV frame, in bytes (1080i50)
int avio_r8(AVIOContext *s)
it s the only field you need to keep assuming you have a context There is some magic you don t need to care about around this just let it vf offset
int flags
A combination of AV_PKT_FLAG values.
static int read_header(FFV1Context *f, RangeCoder *c)
#define i(width, name, range_min, range_max)
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
char * av_timecode_make_smpte_tc_string2(char *buf, AVRational rate, uint32_t tcsmpte, int prevent_df, int skip_field)
Get the timecode string from the SMPTE timecode format.
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
static av_always_inline AVRational av_inv_q(AVRational q)
Invert a rational.
int64_t avio_seek(AVIOContext *s, int64_t offset, int whence)
fseek() equivalent for AVIOContext.
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return or at least make progress towards producing a frame
uint8_t * buf
actual buffer data
#define AVIO_SEEKABLE_NORMAL
Seeking works like for a local file.
int avio_read(AVIOContext *s, unsigned char *buf, int size)
Read size bytes from AVIOContext into buf.
IDirect3DDxgiInterfaceAccess _COM_Outptr_ void ** p
static int read_probe(const AVProbeData *p)
const uint8_t(* audio_shuffle)[9]
This structure stores compressed data.
int av_dict_set(AVDictionary **pm, const char *key, const char *value, int flags)
Set the given entry in *pm, overwriting an existing entry.
int64_t pos
byte position in stream, -1 if unknown
int64_t bit_rate
The average bitrate of the encoded data (in bits per second).
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
int ffio_read_size(AVIOContext *s, unsigned char *buf, int size)
Read size bytes from AVIOContext into buf.
int avio_feof(AVIOContext *s)
Similar to feof() but also returns nonzero on read errors.